[asterisk-dev] Codec Framing/Packetization limits

[James Cloos]

>>>>> "Dan" == Dan Austin <Dan_Austin at Phoenix.com> writes:

Dan> Any comments or suggestions about any of the values would be
Dan> appreciated, but I am most interested in what the Max values
Dan> should be for each codec.

Everything looks sane except for (most of) the max values.

(And that there should be rows for both ilbc-30ms and ilbc-20ms.)

The only real max value should be the number of codec blocks that fit
with rtp, srtp, udp/sctp/dccp, ipsec, ip, vlan, frame, et al overhead
into a 1500 octet frame, since there will be uses where the roundtrip
delay is less of an inconvenience than using even just a bit more bw.

And it shouldn't matter whether multiple codec blocks arrive all in
one single rtp packet or split across several rtp packets, as long
as no block is split between packets.

That said, for most uses the sane maximums will of course be closer
to the point where the audio delay becomes noticeable, but those,
tighter, settings are best done in the config files rather than in
the source.

It looks like slinear is set to fit into a 1280 octet ip packet.
That (1120) order of magnitude would be a reasonable maximum octet
size for the rest of the codecs.

-JimC
-- 
James Cloos <cloos at jhcloos.com>         OpenPGP: 0xED7DAEA6